[Asterisk-Users] Problem with Cisco local conference and hangup

[Leif Neland]

> On Nov 14, 2005, at 5:27 PM, C F wrote:
>> On 11/14/05, Leif Neland <leifn at neland.dk> wrote:
>>> ---- Original Message ----
>>> From: "C F" <shmaltz at gmail.com>
>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>> <asterisk-users at lists.digium.com> Sent: Monday, November 14, 2005
>>> 4:50
>>> PM Subject: [Asterisk-Users] Problem with Cisco local conference and
>>> hangup
>>>
>>>> Cisco 7960 gets a call from zap/1, hits conf to call out on zap/2,
>>>> then hits join, after a while cisco hangsup, at which point zap/1
>>>> and
>>>> zap/2 can still talk, shouldn't asterisk hangup on all three?
>>>
>>> That is the way I would prefer it to work.
>>> Like an attended transfer.
>>
>> I cannot understand why, why not use attended transfer then?
>
> Because the person performing the conference can speak to both
> parties at the same time before hanging up and "transferring" the
> call. With an attended transfer you can only speak to one caller or
> the other, not both.
>
> Also, it is worthy to note that this feature, as well as transfers to
> outside numbers, creates a toll-fraud concern. Here's the scenario:
>
> 1.) Employee's wife calls employee at work - Local call.
> 2.) Employee conferences/transfers wife to inlaws in New Zealand -
> Long distance call.
> 3.) Upshot: The company ends up paying the long distance for the wife
> to speak to her mother.
>
> (Of course, if your company is in NZ, this wouldn't be a big deal,
> but it is for everyone else!)
>
So it should act differently if the outgoing call is internal or external.
Or outgoing calls should be blocked.
Or outgoing calls which are not tollfree should be blocked. (I have free 
calls to all the subscribers of the same voisp as me)

Is there support in SIP for the voisp supplying the cost when setting up the 
call?

Leif